With the development of data communications and multimedia service requirements, fourth generation mobile communications meeting operation needs of mobile data, mobile computing, and mobile multimedia initiates to emerge. The fourth generation mobile communication technology (4G) includes two standards: time division long term evolution (TD-LTE) and frequency-division duplex long term evolution (FDD-LTE). 4G integrates 3G and WLAN, and can be used to perform fast transmission of data, a high quality audio, a high quality video, a high quality image, and the like.
Voice over LTE (VoLTE) is an LTE voice solution based on an IP multimedia subsystem (IMS) network. The VoLTE is essentially different from a 2G or 3G audio call. The VoLTE is an end-to-end voice solution based on a 4G network on an all-IP condition.
FIG. 1 is a signaling interaction diagram of an existing VoLTE call flow. As shown in FIG. 1, in a call process, user equipment (UE) sends an INVITE message to a proxy-call session control function (P-CSCF)/session border controller (SBC) by using Session Initiation Protocol (SIP) signaling. The INVITE message includes codec information. For adaptive multirate narrowband (AMR-NB) and adaptive multirate wideband (AMR-WB) codecs, the message carries rate set information supported by the UE. An IMS sends the INVITE message to a peer end, that is, sends codec information to the peer end, and performs voice bearer plane codec negotiation by using the SIP signaling. Then, the P-CSCF/SBC receives a response message 180 fed back by the peer end, and the response message 180 carries bearer plane codec information returned by the peer end. The P-CSCF/SBC sends the response message 180 to the UE. In this way, by using SIP signaling negotiation, the UE, the P-CSCF/SBC, and the peer end learn codec information used for a current call. If the codec information is an AMR-NB or AMR-WB codec, the rate set information is further obtained. Subsequently, after the call is connected, bearer plane interaction is performed according to a result of the SIP signaling negotiation. In the foregoing process, an evolved NodeB (eNB) and a serving gateway (S-GW)/PDN-Gateway (P-GW) transfer signaling and bearer information. A signaling and bearer GPRS tunneling protocol (GTP tunnel) tunnel is established between the eNB and the S-GW/P-GW. When receiving an uplink packet sent by the UE, the eNB transfers the uplink packet to the S-GW/P-GW by using a GTP tunnel, and the S-GW/P-GW sends the uplink packet to the P-CSCF/SBC. When sending a downlink packet to the UE, the P-CSCF/SBC first sends the downlink packet to the S-GW/P-GW, the S-GW/P-GW sends the downlink packet to the eNB by using a GTP tunnel, and the eNB transfers the downlink packet to the UE by using an air interface.
In the foregoing all-IP voice solution, the eNB and the S-GW/P-GW are responsible only for transferring an IP packet (including signaling and a bearer). In the foregoing all-IP voice solution, a service rate cannot be dynamically adjusted according to transmission quality of an air interface of the eNB. This results in a problem of a packet loss, a long delay, or a low resource utilization rate caused by a mismatch between an actual transmission capability of the eNB and the service rate.